This invention relates to digital audio signal processing, specifically to an improved method for imparting a harmonic distortion effect to a digital audio signal.
Most electronic systems that process (amplify, transmit, etc.) analog signals are designed to minimize the amount of harmonic distortion added to the signal. When processing audio signals, however, some types of harmonic distortion are desirable. Audio professionals, musicians, and audiophiles widely pursue the sonic characteristics of vacuum tubes, analog recording tape, transducers, and a variety of solid-state distortion devices. Harmonic distortion due to non-linear transfer characteristics, either inherent or deliberately induced, is mainly responsible for imparting the desirable effect.
The progress of digital recording and signal processing technologies is at a point where most audio signal processing tasks can be performed satisfactorily in the digital domain. One notable exception is distortion effects. Digital distortion effects generally lack the accuracy needed to be convincing replacements for their analog counterparts and tend to possess a harsh quality that reveals their digital origins.
U.S. Pat. No. 4,995,084 to Pritchard (Feb. 19, 1991) relates to a variety of analog semiconductor circuits for emulating the general characteristics of a vacuum tube amplifier. It also discloses digital versions that approximate the operation of the analog circuits.
U.S. Pat. No. 5,570,424 to Araya et al. (Oct. 29, 1996) relates to fixed distortion algorithms based on higher degree expressions, mainly comprising a series of cubic expressions.
Extending the Karpus-Strong Algorithm to Synthesize Electric Guitar Timbres with Distortion and Feedback by Charles R. Sullivan in the Computer Music Journal (Fall 1990) relates to an algorithm for computer music synthesis that includes a simple distortion function for providing a soft clipping effect.
The distortion generation methods used by Pritchard, Araya et al., and Sullivan use single-purpose, fixed algorithms that offer no programmability or variation of the distortion effect other than setting the input signal level. They do not emulate the distortion characteristics of any actual analog distortion device but instead provide only some characteristics of a generalized class of distortion device. They also do not provide any method of suppressing the aliasing noise that can result when performing a non-linear operation in a digital system.
Cool Edit Software version 1.53 by Syntrillium Software Corporation (1992-1996) relates to a computer program that provides numerous utilities for modifying digital audio files. It includes a distortion utility for graphically building a distortion curve from straight line segments. The curve represents the graph of an arbitrary transfer function that is applied symmetrically to the positive and negative values of an audio file. The term transfer function refers to the static or direct-current (DC) transfer characteristics of a distortion device. Cool Edit""s distortion is limited to generating only odd harmonics due to the symmetry of the transfer function. It does not provide any method of suppressing the aliasing noise that can result when performing a non-linear operation in a digital system. The segmentation of the distortion curve generates high harmonics that are usually undesirable and further add to the aliasing noise.
SPKitWaveShaperxe2x80x94User""s Guide by Kai Lassfolk in Sound Processing Kitxe2x80x94User""s Guide (1995) relates to a distortion routine included in a unified collection of computer software routines designed for modifying digital audio files. The distortion routine imports a user-supplied file containing look-up table values that it uses to non-linearly transform an audio file. The routine simply executes the distortion process and has no bearing on the nature and origin of the distortion""s characteristics. It does not provide any method of suppressing the aliasing noise that can result when performing a non-linear operation in a digital system.
Digital Waveshaping Synthesis by Marc Le Brun and Digital Synthesis of Complex Spectra by Means of Multiplication of Nonlinear Distorted Sine Waves by Daniel Arfib, both in the Journal of the Audio Engineering Society (April 1979 and October 1979, respectively) relate to a digital music synthesis technique for generating a complex waveform by applying a non-linear waveshaping function to a sinusoidal waveform. The work of Le Brun and Arfib provides a good theoretical insight into the relationship of a waveshaping function and the harmonics it produces. However, the focus is on the direct synthesis of musical instrument sounds and, as such, offers no method for applying a distortion effect to digital audio signals.
Musical Applications of Microprocessorsxe2x80x94Second Edition by Hal Chamberlin (1985) relates to music synthesis and sound modification, providing a comprehensive collection of techniques, both analog and digital. More specifically, pages 48-54 describe the application of non-linear waveshaping to instrumental signals using analog techniques. Pages 473-480 describe digital non-linear waveshaping and state the advantages of digital waveshaping over the analog techniques described in the earlier section. The digital waveshaping functions described are limited to the synthesis of musical instrument sounds when applied to a sinusoidal waveform. No method is offered for generating a waveshaping function that would emulate another distortion device when applied to a digital audio signal.
U.S. Pat. Nos. 4,868,869 and 4,991,218 to Kramer (Sep. 19, 1989 and Feb. 5, 1991) relate to the non-linear waveshaping of digital audio signals, specifically when applied with a look-up table and performed in real time. They describe various general methods for generating waveshaping functions that are suitable primarily for experimentation. None involve the analysis of another distortion device. As such, no method is offered for generating a waveshaping function that would emulate another distortion device. No method is provided for suppressing the aliasing noise that can result when performing a non-linear operation in a digital system.
U.S. Pat. No. 5,235,410 to Hurley (Aug. 10, 1993), U.S. Pat. No. 4,949,177 to Bannister et al. (Aug. 14, 1990), U.S. Pat. No. 5,253,043 to Gibson (Oct. 12, 1993), and U.S. Pat. No. 5,349,546 to Sharman (Sep. 20, 1994) all relate to digital video signal processing, specifically non-linear processing that includes methods for reducing alias frequencies in the output signal. Although they do apply a form of distortion to a digital signal, their non-linear processing is specifically for video and is irrelevant for audio.
Besides the limitations already stated, none of the prior art perform direct analyses of existing distortion devices for deriving methods or models that can be used to emulate the distortion devices.
Accordingly, several objects and advantages of my invention are to provide a distortion modeling and synthesis method whereby analog audio distortion devices are accurately emulated using digital techniques, to provide a distortion modeling and synthesis method capable of synthesizing a virtually unlimited variety of distortion effects, and to provide a distortion modeling and synthesis method capable of suppressing the aliasing noise that can result when performing a non-linear operation in a digital system.
It is an additional object of the invention to provide a distortion modeling and synthesis method that permits modification of the distortion in terms of the harmonics it produces.
It is an additional object of the invention to provide a distortion modeling and synthesis method that facilitates efficient storage of distortion models for later recall and usage.
To accomplish the above-mentioned objects, the harmonic distortion effect of an audio distortion device is modeled by determining the harmonic characteristics of the distortion imparted to a signal by the distortion device. A distortion synthesizer uses the resulting distortion model to synthesize the distortion characteristics of the distortion device. The synthesized harmonic distortion effect imparted to a signal is virtually identical to that of the distortion device used to create the model.
To create a distortion model, the distortion device is operated using a pure, undistorted sinusoidal signal as an input. Spectral analysis is performed on the output signal from the distortion device to determine the amplitude and phase of the signal""s harmonic components. The harmonic amplitude and phase parameters form the distortion model. A process is provided for correcting the phase parameters of distortion models created from distortion devices that induce phase shifts into the distorted signal. The distortion synthesizer provides a mechanism that uses the distortion model in performing a non-linear transformation of a digital signal. More precisely, the distortion synthesizer uses a mathematical function that accepts the harmonic amplitude and phase parameters of the distortion model. Applying the function to a digital signal of interest imparts harmonic distortion to the signal according to the distortion model.
Several advantages are realized by the modeling and synthesis approach to digital distortion as implemented in my invention. Foremost is the high accuracy with which the non-linearity of a distortion device is reproduced. Another advantage is the versatility of the invention. Rather than being a method for synthesizing the effect of one specific type of distortion device, the present invention provides a general method capable of synthesizing the effects of different types of distortion devices. The modeling method can extract the static non-linear transfer characteristics of a device even if its non-linear element can only be accessed dynamically, such as with analog recording tape. A further advantage is that the distortion model""s components directly relate to the way humans perceive sound; that is, by the sound""s harmonic content. As such, the distortion model can be modified to alter its sonic characteristics predictably. A distortion model contains a relatively small amount of data, typically less than one hundred numeric values, allowing for efficient storage of multiple models in a data-storage device.
A problem that occurs when generating harmonic distortion digitally is the production of aliasing noise. The frequencies of some of the harmonics added to a signal by the non-linear distortion process may exceed one-half of the sampling rate. Consequently, those harmonics fold back into the audible signal as aliasing noise. The ability to modify the distortion model of my invention has the further advantage of making it possible to reduce aliasing noise by eliminating the upper harmonics from the model. The distortion synthesizer contains further enhancements specifically for controlling aliasing noise. A frequency bandsplitter separates the input signal into a low frequency signal and a high frequency signal. Distortion is applied only to the low frequency signal, which is subsequently recombined with the high frequency signal. The bandlimiting action of the bandsplitter creates spectral headroom that permits the addition of harmonics without causing aliasing noise. To apply distortion to the full bandwidth of the input signal without causing aliasing noise, sampling-rate conversion is used. The distortion synthesizer provides mechanisms for increasing and decreasing the sampling rates of digital signals. Before applying the distortion, the input signal is converted to a higher sampling rate to create spectral headroom for accommodating the added harmonics. After applying the distortion, the signal is converted back to the original sampling rate. The conversion process removes the added harmonics having frequencies that exceed one-half of the reduced sampling rate, thereby preventing aliasing noise. Combining the bandsplitter with sampling-rate conversion further extends the spectral headroom.